Nobody hasn't a sollution for me ?
- Original Message -
From: Etann
To: asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2012 5:20 PM
Subject: [asterisk-users] Answering machine dectection (AMD)
Hi,
I'll try to havebeen help for asterisk AMD module.
I need to add a semi-colon to a variable, but no matter how I quote
it, the parser ignores it and considers the semi-colon as the
beginning of a comment.
Si how do I concatenate the content of a variable to a semi-colon? I
tried surrounding it with double quotes, single quotes, using a
backslash
Hello!
I do realize that this is not strictly an Asterisk question, but I have a
feeling that the knowledge might be found here.
Does anyone have a good set of regional settings (rings/dial tones/cadences)
for the SPA-3000 for Sweden?
Thanks in advance!
--
Noone knows that? Where/whom could I ask?
Thanks
Il 10/02/2012 12:30, Matteo Fortini ha scritto:
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe()
I was using 1.8.8.1 and now upgraded it to 1.8.9.1. Here is a problem I
have with Asterisk logging if someone can point me to the right direction.
With allowguest=no, Asterisk 1.8.9.1 doesn't create anything in the full
log so my fail2ban can't ban the unregistered call attempt on my server.
How
Hi everyone,
Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about
5000 numbers and then put the call to agents right away and pull up the CRM
based on the number dialed. So, I am going to be doing some PHP+Ajax work.
I am familiar with spool files but I don't like the fact
On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote:
Hi everyone,
Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about
5000 numbers and then put the call to agents right away and pull up the CRM
based on the number dialed. So, I am going to be
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port = $ast_agi-get_variable(SIPPEER($jkh,port));
Thanks for the input but using spool files or AMI or AGI is way different
from each other and that is what I want to get an input on. I do have hands
on with all methods like I noted but want to know what the trend is
now-a-days and what is more robust and proven out of all three.
Best,
On Sat,
On 02/11/2012 04:41 AM, CDR wrote:
I need to add a semi-colon to a variable, but no matter how I quote
it, the parser ignores it and considers the semi-colon as the
beginning of a comment.
Si how do I concatenate the content of a variable to a semi-colon? I
tried surrounding it with double
I'd definitely go with AMI !
On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs asteriskcod...@gmail.comwrote:
Thanks for the input but using spool files or AMI or AGI is way different
from each other and that is what I want to get an input on. I do have hands
on with all methods like I noted but
On 11 Feb 2012, at 13:41, Kevin P. Fleming wrote:
At this time, there's no way to do it directly in the dialplan
In extensions.conf
[globals]
SEMICOLON=;
Then use ${SEMICOLON}
S--
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Sammy,
Would you care to elaborate please. Have you had experience doing such a
campaign using AMI? Maybe you can share of the code.
Most appreciated,
On Sat, Feb 11, 2012 at 10:15 AM, Sammy Govind govoi...@gmail.com wrote:
I'd definitely go with AMI !
On Sat, Feb 11, 2012 at 6:39 PM,
zaharova.t2011 zaharova.t2011 http://ogcoatings.com/images/ber.html reyd
reyd
skiminok-ral 589632147
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New
I just set up a WRT54GS and now I can't dial out or in.
sip show registry shows:
CODE: SELECT ALL
Hostdnsmgr Username Refresh State
Reg.Time
draytel.org:5060N x 120 Request Sent
I
Hi,
in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an
Audiocodes.
I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems
to say so as well, but I
want to make sure, and fixing the Audiocodes is not an option in this
particular case - don't ask.
Can
On (16:48 11/02/12), David Woodfall d...@dawoodfall.net put forth the
proposition:
I just set up a WRT54GS and now I can't dial out or in.
sip show registry shows:
CODE: SELECT ALL
Hostdnsmgr Username Refresh State
Reg.Time
On (17:38 11/02/12), David Woodfall d...@dawoodfall.net put forth the
proposition:
On (16:48 11/02/12), David Woodfall d...@dawoodfall.net put forth the
proposition:
I just set up a WRT54GS and now I can't dial out or in.
sip show registry shows:
CODE: SELECT ALL
Host
Sammy Govind wrote:
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port =
Linksys firmware?
I've had issues with older firmwares and VoIP
Sent from my iPhone 4S
On Feb 11, 2012, at 1:05 PM, David Woodfall d...@dawoodfall.net wrote:
On (17:38 11/02/12), David Woodfall d...@dawoodfall.net put forth the
proposition:
On (16:48 11/02/12), David Woodfall
On (14:02 11/02/12), Robert-IPhone rhuddles...@gmail.com put forth the
proposition:
Linksys firmware?
I've had issues with older firmwares and VoIP
Yes the Linksys firmware. I'm not totally sure whether to put DD-WRT
on it at the moment. Thinking I might buy something newer instead.
Sent
I seem to recall 3.3.2 had other problems (for me at least) so I only used it
on some phones. Their 3.3 branch is at 3.3.4 now though, I would try that.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim DeVito
Sent:
I've been lurking on the dev discussion on creating nat=auto. It all
leads me to think there's no reason to use nat=no.
We have about 60 internal sip extensions connected to an multihomed
asterisk box where the external ip is not nat'ed. Each of the internal
sip contexts has nat=no. On
If your server is open to the internet and in SIP general section you have
nat=no and in peers you have nat=yes or vice versa then it's possible to
enumerate your extension. Because Asterisk responds with different messages
if the extension exists or not based on that difference in the nat setting
This is my first 1.8 install. I'm trying to set up CEL and am getting the
following errors. Does anyone have any ideas on where to look? res_odbc and
cdr_adaptive_odbc appear to be working, the CDRs are working.
[Feb 11 23:44:36] WARNING[23125]: cel_odbc.c:699 odbc_log: Column type -9
Hey Ron,
Thanks for taking out time for this weird issue. No this is the only code
thats running and I simply copy pasted it here. I'll go through the artivle
you mentioned and other advices you gave may hopefully resolve this issue.
But in general its beyond my logic to see whats actually going
Yes why not,
I made an aut-odialer (the code I can share on my blogpost in couple of
days for you.) The basic structure of the script/code was to:
1- Start, connect to DB, fetch campaign data
2- Fetch numbers to dial from campaign, If no numbers goto step 6
3- Feed those number in a loop to AMI
Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files
that come with new firmware. I have seen wiered things by using olde cfg
files
On Friday, February 10, 2012, Mike l...@net-wall.com wrote:
Hi,
I just moved many Polycom phones from
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